audio_core: remove time stretcher
Also drop the SoundTouch dependency
This commit is contained in:
Andrea Pappacoda
parent
550844e5e8
commit
faf6a9876c
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@ -7,9 +7,6 @@
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[submodule "dynarmic"]
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path = externals/dynarmic
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url = https://github.com/MerryMage/dynarmic.git
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[submodule "soundtouch"]
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path = externals/soundtouch
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url = https://github.com/citra-emu/ext-soundtouch.git
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[submodule "libressl"]
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path = externals/libressl
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url = https://github.com/citra-emu/ext-libressl-portable.git
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@ -68,9 +68,6 @@ if (YUZU_USE_EXTERNAL_SDL2)
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add_library(SDL2 ALIAS SDL2-static)
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endif()
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# SoundTouch
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add_subdirectory(soundtouch)
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# Cubeb
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if(ENABLE_CUBEB)
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set(BUILD_TESTS OFF CACHE BOOL "")
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@ -1 +0,0 @@
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Subproject commit 060181eaf273180d3a7e87349895bd0cb6ccbf4a
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@ -36,8 +36,6 @@ add_library(audio_core STATIC
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splitter_context.h
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stream.cpp
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stream.h
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time_stretch.cpp
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time_stretch.h
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voice_context.cpp
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voice_context.h
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@ -63,7 +61,6 @@ if (NOT MSVC)
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endif()
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target_link_libraries(audio_core PUBLIC common core)
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target_link_libraries(audio_core PRIVATE SoundTouch)
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if(ENABLE_CUBEB)
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target_link_libraries(audio_core PRIVATE cubeb)
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@ -7,7 +7,6 @@
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#include <cstring>
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#include "audio_core/cubeb_sink.h"
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#include "audio_core/stream.h"
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#include "audio_core/time_stretch.h"
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#include "common/assert.h"
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#include "common/logging/log.h"
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#include "common/ring_buffer.h"
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@ -23,8 +22,7 @@ class CubebSinkStream final : public SinkStream {
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public:
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CubebSinkStream(cubeb* ctx_, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
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const std::string& name)
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: ctx{ctx_}, num_channels{std::min(num_channels_, 6u)}, time_stretch{sample_rate,
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num_channels} {
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: ctx{ctx_}, num_channels{std::min(num_channels_, 6u)} {
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cubeb_stream_params params{};
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params.rate = sample_rate;
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@ -131,7 +129,6 @@ private:
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Common::RingBuffer<s16, 0x10000> queue;
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std::array<s16, 2> last_frame{};
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std::atomic<bool> should_flush{};
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TimeStretcher time_stretch;
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static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
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void* output_buffer, long num_frames);
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@ -205,25 +202,7 @@ long CubebSinkStream::DataCallback([[maybe_unused]] cubeb_stream* stream, void*
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const std::size_t num_channels = impl->GetNumChannels();
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const std::size_t samples_to_write = num_channels * num_frames;
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std::size_t samples_written;
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/*
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if (Settings::values.enable_audio_stretching.GetValue()) {
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const std::vector<s16> in{impl->queue.Pop()};
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const std::size_t num_in{in.size() / num_channels};
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s16* const out{reinterpret_cast<s16*>(buffer)};
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const std::size_t out_frames =
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impl->time_stretch.Process(in.data(), num_in, out, num_frames);
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samples_written = out_frames * num_channels;
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if (impl->should_flush) {
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impl->time_stretch.Flush();
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impl->should_flush = false;
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}
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} else {
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samples_written = impl->queue.Pop(buffer, samples_to_write);
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}*/
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samples_written = impl->queue.Pop(buffer, samples_to_write);
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const std::size_t samples_written = impl->queue.Pop(buffer, samples_to_write);
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if (samples_written >= num_channels) {
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std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),
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@ -7,7 +7,6 @@
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#include <cstring>
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#include "audio_core/sdl2_sink.h"
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#include "audio_core/stream.h"
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#include "audio_core/time_stretch.h"
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#include "common/assert.h"
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#include "common/logging/log.h"
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//#include "common/settings.h"
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@ -27,7 +26,7 @@ namespace AudioCore {
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class SDLSinkStream final : public SinkStream {
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public:
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SDLSinkStream(u32 sample_rate, u32 num_channels_, const std::string& output_device)
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: num_channels{std::min(num_channels_, 6u)}, time_stretch{sample_rate, num_channels} {
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: num_channels{std::min(num_channels_, 6u)} {
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SDL_AudioSpec spec;
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spec.freq = sample_rate;
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@ -116,7 +115,6 @@ private:
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SDL_AudioDeviceID dev = 0;
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u32 num_channels{};
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std::atomic<bool> should_flush{};
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TimeStretcher time_stretch;
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};
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SDLSink::SDLSink(std::string_view target_device_name) {
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@ -1,68 +0,0 @@
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// Copyright 2018 yuzu Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <algorithm>
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#include <cmath>
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#include <cstddef>
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#include "audio_core/time_stretch.h"
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#include "common/logging/log.h"
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namespace AudioCore {
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TimeStretcher::TimeStretcher(u32 sample_rate, u32 channel_count) : m_sample_rate{sample_rate} {
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m_sound_touch.setChannels(channel_count);
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m_sound_touch.setSampleRate(sample_rate);
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m_sound_touch.setPitch(1.0);
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m_sound_touch.setTempo(1.0);
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}
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void TimeStretcher::Clear() {
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m_sound_touch.clear();
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}
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void TimeStretcher::Flush() {
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m_sound_touch.flush();
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}
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std::size_t TimeStretcher::Process(const s16* in, std::size_t num_in, s16* out,
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std::size_t num_out) {
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const double time_delta = static_cast<double>(num_out) / m_sample_rate; // seconds
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// We were given actual_samples number of samples, and num_samples were requested from us.
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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const double max_latency = 0.25; // seconds
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const double max_backlog = m_sample_rate * max_latency;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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if (backlog_fullness > 4.0) {
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// Too many samples in backlog: Don't push anymore on
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num_in = 0;
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}
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// We ideally want the backlog to be about 50% full.
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// This gives some headroom both ways to prevent underflow and overflow.
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// We tweak current_ratio to encourage this.
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constexpr double tweak_time_scale = 0.05; // seconds
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const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
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current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0);
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// This low-pass filter smoothes out variance in the calculated stretch ratio.
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// The time-scale determines how responsive this filter is.
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constexpr double lpf_time_scale = 0.712; // seconds
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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// Place a lower limit of 5% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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m_stretch_ratio = std::max(m_stretch_ratio, 0.05);
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m_sound_touch.setTempo(m_stretch_ratio);
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LOG_TRACE(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, m_stretch_ratio,
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backlog_fullness);
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m_sound_touch.putSamples(in, static_cast<u32>(num_in));
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return m_sound_touch.receiveSamples(out, static_cast<u32>(num_out));
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}
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} // namespace AudioCore
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@ -1,34 +0,0 @@
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// Copyright 2018 yuzu Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#pragma once
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#include <cstddef>
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#include <SoundTouch.h>
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#include "common/common_types.h"
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namespace AudioCore {
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class TimeStretcher {
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public:
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TimeStretcher(u32 sample_rate, u32 channel_count);
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/// @param in Input sample buffer
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/// @param num_in Number of input frames in `in`
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/// @param out Output sample buffer
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/// @param num_out Desired number of output frames in `out`
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/// @returns Actual number of frames written to `out`
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std::size_t Process(const s16* in, std::size_t num_in, s16* out, std::size_t num_out);
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void Clear();
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void Flush();
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private:
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u32 m_sample_rate;
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soundtouch::SoundTouch m_sound_touch;
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double m_stretch_ratio = 1.0;
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};
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} // namespace AudioCore
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@ -342,12 +342,6 @@ fps_cap =
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# null: No audio output
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output_engine =
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# Whether or not to enable the audio-stretching post-processing effect.
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# This effect adjusts audio speed to match emulation speed and helps prevent audio stutter,
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# at the cost of increasing audio latency.
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# 0: No, 1 (default): Yes
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enable_audio_stretching =
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# Which audio device to use.
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# auto (default): Auto-select
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output_device =
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